similar to: Stereo voice not being retained

Hey Kevin! Once you start dropping under (I think) 32kbps, the codec naturally starts losing stereo separation for the benefit of quality. If you really want perfect stereo at 16kbps CBR (this is horrifically low in my opinion) you might be better off splitting the audio and encoding each part at 8kbps CBR. Or, there may be a magic "enforce 100% stereo" one could pass? (doesn't look

Hi, I have downloaded the latest stable version *1.0.2 opus audio codec* from the following link: http://www.opus-codec.org/downloads/ *Details of OS & CPU:* OS : Microsoft Windows XP CPU : intel core 2 Duo cpu I am using Microsoft visual C++ 2010 Express to build the codec (opus.vcxproj available in the package downloaded). I am able to build it & could generate the application

Hi I used opus_demo to clarify how opus works, I try this command: opus_demo -e" audio 8000 1 8000 -cbr /somedirectory/pcm_mono_ochomil.raw /somedirectory/test.opus And I get the file test.opus, then comes the first problem, this file hasn't header so i can't reproduce it using an external tool, how can i add a header to this file Then when i trie to run opus_demo -d 8000 1

Hi I'm having a problem compiling opus-tools-0.1.7. Version opus-tools-0.1.6 seems to compile OK. I've tried with opus-1.0.3 and opus-1.1-beta. The errors are like this:- "undefined reference to `sqrtf'" etc. This OS is Peppermint Three, similar to Ubuntu 12.04. It uses:- gcc --version gcc (Ubuntu/Linaro 4.6.3-1ubuntu5) 4.6.3 Google says it's maybe something to do

Hi, Dear alls I had encounted a problem in using opus-tool-0.2-opus-1.3(download from www.opus-codec.org, Win64 binaries: opus-tools-0.2-win64.zip). First, I use opusenc to generate a opus file which input is a 9-channels wav file. The opusenc can work. Then, I use opusdec to transform the generated opus file to wav file, but opusdec can't work. I received the error message: failed to open

Hi, I have found differences in quality between opus_demo and opusenc/opusdec. I used for both applications the same raw pcm file,16 bit,48khz,litle endian. i use libopus 1.1-beta and opus-tools-0.1.7. The command for opus_demo is: opus_demo audio 48000 1 64000 -cvbr -framesize 10 in.pcm out.pcm For opusenc/dec: opusenc --raw --raw-chan 1 bitrate 64 -cvbr --framesize 10 in.pcm in.opus

Hi, I've been using the command line opusenc/opusdec tools for a while now to encode/decode files. The next version of OS X changes a few things, so I was looking to recompile these tools using a version of the OS X SDK > 10.7 (needed for apple's "hardened runtime") but had no luck. How can I compile a static version of opus tools on OS X that doesn't depend on any other

opusdec -V opusdec opus-tools f2a2e88 (using libopus unknown) I've got an opus file encoded from a .wav off a cd, 44100Hz: opusinfo 2-24-Overture_in_C_\(In_Memoriam\).opus Processing file "2-24-Overture_in_C_(In_Memoriam).opus"... New logical stream (#1, serial: 38134f1f): type opus Encoded with libopus unknown User comments section follows... ENCODER=opusenc from opus-tools

opus_decode() produces 16-bit signed linear PCM, and opus_decode_float() produces 32-bit floating point PCM that is useful when you want a higher bit depth. If you need 8-bit linear PCM then a simple solution would be to use only the top 8 bits of each 16-bit sample from opus_decode(). Note that the WAV format uses unsigned rather than signed integers for 8-bit linear PCM. (It uses signed for

Hi I am an Audacity (http://audacity.sourceforge.net/) developer and I'd like to see Audacity reading and writing Opus files as easily as it does others. What would you recommend as a way forward for this? I am thinking that (Win) we should ship opusenc.exe and opusdec.exe and send command line stuff to them; that way we wouldn't need to dip into your code and as you update we would

Hi All, First time poster to this group, please ignore my ignorance? I?m trying to use Opus 1.1 on a 400MHz MIPS 24k CPU (AR9331, specifically, like in the Arduino Yun). I?ve successfully built (I think) opus-1.1 and opus-tools-1.8 and they run, but are dog slow. opus-1.1 does have the ?enable-fixed-point option set, as this chip only has soft-float. My short test file (less than one

Dear authors of Opus, currently, the manpages that come with opus-tools are written in the traditional man(7) markup language. I am proposing to rewrite them into the semantic markup of the mdoc(7) language. I am willing to do the work. Both the man(7) and mdoc(7) languages have been around for decades, and are supported by the prevalent formatters: groff(1) on most Linuxes and mandoc(1) on the

Hello Ralph, > Likewise opus_encode() takes 16 bit samples, so you need to extend each > sample from an 8 bit source before encoding. Two questions 1. In opusenc.c which API does the extending the 8-bit to 16-bit? 2. If that is the case then how will 24 bit PCM sample work? Regards Amit On Thu, Jan 7, 2016 at 12:21 PM, Ralph Giles <giles at thaumas.net> wrote: > On 07/01/16

using iTunes i've noticed that AAC is very good at re-encoding own lossy sound. let's test Opus! neroaacenc.exe -q 0.75 -if 000.wav -of 001.m4a neroaacdec.exe -if 001.m4a -of aac001.wav wavdiff.exe 000.wav aac001.wav Comparing 000.wav - aac001.wav... Max diff: -17.3867dB RMS diff: -33.0851dB Mean diff: -32.4582dB opusenc.exe --bitrate 512 "000.wav" 001.opus opusdec.exe 001.opus

Hello Mark, The resulting 8 bit file has a lot of squelching noise compared to the 16 bit output from OPUS decoder. During encode I am using popi16fmtBuffer[ui32Loop] = (opus_int16)pcRdBuf[ui32Loop]; And during decode since the data is in the lower 8 bit I use pc8bitSamples[ui32Loop] = ((unsigned short)pcop16OutBuf[ui32Loop] ^ 0x80); Regards Amit On Sat, Jan 9, 2016 at 2:39 PM, Amit Ashara

Good morning. I've ran a test against MP3 format. Code: (first convert tested audio file to 16 bit 48khz with sox.exe if needed) lame.exe -b 320 48khzfilein.wav -o fileout.mp3 lame --decode fileout.mp3 -o fileout.mp3.wav opusenc.exe --bitrate 320 48khzfilein.wav fileout.opus opusdec.exe fileout.opus fileout.opus.wav wavdiff.exe 48khzfilein.wav fileout.mp3.wav -diff fileout.mp3.delta.wav

hello all - since i am such a newbie, i thought i would be the ideal person to put together a guide on how i got the opus tools working. ?i am using virtualbox and a new centOS-linux virtual-machine. i chose to use the minimalist centOS iso file so there would be no pre-installed packages. any suggestions or improvements to this guide would be most appreciated. ?? also, wish to express my

>On 11/20/2013 03:37 PM, O'Connor, Kevin wrote: >> I have an application I intend to use Opus for that involves creating >> recordings of two-party conversations where each party is saved as a >> separate channel. Audio may be later processed or analyzed on a >> per-party basis so if audio in one channel affects the other channel, it >> could create problems.

>>> Agustín Dall'Alba <agustin at dallalba.com.ar> schrieb am 14.04.2017 um 22:53 in Nachricht <CAHBqS-w3v44WM5x+_4XdFMkD42A2iYTbEWKEBmvJc2P3Y-LJGA at mail.gmail.com>: > I halved the volume of the sample before encoding with > `sox -v 0.5 floex.wav quiet.wav` and now I can't ABX it succesfully anymore. > So the artifact I heard was just clipping when encoding

Jean-Mark sarkasm. Jean-Markasm. (Bonus points for providing an actual noisy WAV! ^_^) On 30/10/2017 20:28, Jean-Marc Valin wrote: Hi, Before I comment on the graphics you posted to visualize the difference between two audio signals, I'd like to ask for your help in evaluating my JPEG encoder. I've encoded an image with JPEG and then computed the difference with the original. I then